The present invention pertains in general to analog-to-digital converters, and more particularly, to the use of the digital filter section to process and store digital audio on a digital storage media.
Analog-to-digital converters have seen increased use in the audio industry. Due to the increased level of sophistication in the processing of analog information, digital techniques have been utilized to process this analog information. By converting the analog signal into a digital signal and utilizing available digital processing techniques, a higher degree of versatility is provided to the user. This digital processing is utilized to process the information in order to provide various types of outputs after processing. One type of output is a fairly high filtered output that is typically provided by a digital filtering process that requires a very sharp filter response with minimal aliasing. This type of filter, unfortunately, has a significant group delay associated therewith, due to the fact that this type of filter requires a relatively long Finite Impulse Response (FIR) filter. While this is necessary to provide a high quality sound recording, the group delay can present a problem with respect to an artist listening to the soundtrack while it is being recorded. In order to achieve a lower group delay, a much shorter filter with less taps is required, which inherently has a poor filter response.
Other types of processing that can be provided are, for example, a psycho-acoustic filter that shapes the noise response of a given filter output to minimize the noise in the portion of the spectrum associated with the optimum response of the human ear, i.e., approximately 2 KHz, and then increase the noise level above and below that frequency. These types of filters are very useful when a conversion from a high resolution digital output to a low resolution digital output is needed. This noise shaping is directed toward the truncation noise that is related to processing at one word length and then reducing the word length by truncating bits. This filtering does not shape the background noise.
At present, all the above functions require separate processing systems, most of which are not compatible with each other. This presents a disadvantage to the user in that the user must utilize separate systems for the separate functions and is not provided an easy means to facilitate the different systems.
Another aspect of utilizing data conversion devices in audio applications is the requirement for the data conversion device to operate at higher frequencies. Due to the advent of digital storage devices such as the Digital Audio Tape (DAT) and the Digital Video Disk (DVD), the sampling rate of the digital data has been increased, as it is believed that the higher sampling rate allows ultrasonic information to be stored. For example, the sampling rate is now at 96 KHz as compared to previous Compact Disk (CD) formats that have sampling rates slightly above 40 KHz. The result of this is that the fs/2 frequency of 48 KHz is well outside the audio range of most individuals. With the prior CD formats, the fs/2 frequency of slightly greater than 20 KHz could be heard by some individuals.
In order to insure that the noise performance of a 48 KHz sampling rate signal derived from a 96 MU sampling rate signal stored in a digital media was unchanged was to insure that the noise floor at the 96 KHz sampling rate was the same as the noise floor at the 48 KHz sampling rate. Therefore, the bit resolution at the 96 KHz sampling rate had to be maintained as that at the 48 KHz sampling rate, i.e., considerable storage would therefore be required.
The present invention disclosed and claimed herein comprises a method and apparatus for storing data on a digital audio media and then retrieving the audio data in a playback operation. The audio input signal is converted to a digital audio signal at a first sampling rate and at a first word length and having a defined frequency band of operation. This digital audio signal is then noise shaped to shift noise from a low portion of the frequency band thereof to a higher portion of the frequency band thereof. This noise-shaped digital audio signal has a smaller and second word length at a first sampling rate and is then stored on the digital audio media. Thereafter, the stored noise-shaped digital audio signal is retrieved and then converted to the first word length at a second and lower sampling rate. The conversion is done by filtering the data with a digital filter to remove the shifted noise in the higher portion of the frequency band and decimating it to the second sampling rate. This filtered digital audio signal is then input to a digital-to-analog converter operating at the second sampling rate.
In another aspect of the present invention, the noise-shaped digital audio signal can be coupled directly to the digital-to-analog converter, which is operated at the first sampling rate and at the second word length.
In a yet further aspect of the present invention, the step of converting the analog audio signal to a digital audio signal involves processing the signal through an analog modulator and then through a digital decimation filter. The digital filter operates at the first sampling rate but at the second word length. After filtering and decimation, the signal is noise shaped at the first word length prior to the truncation to the second word length, the noise being shaped being the truncation noise. The filter operation is multiplexed, such that the digital filter utilized to filter the output of the analog modulator can be utilized to filter the retrieved converted noise-shaped digital audio signal. Since the digital filter performs a filtering and decimation operation, both steps can be performed therein.